Grandstream Networks HT386 Send Flash Event, Enable Call Features, Use Bell-style, Way Conference

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INFO.

 

 

 

 

 

 

 

 

Send Flash Event

 

Default is NO. If set to yes, flash will be sent as DTMF event.

 

 

 

 

 

 

 

Enable Call Features

 

Default is Yes. Advance call features and feature codes functions are

 

 

 

 

supported locally

 

 

 

 

 

 

 

 

Use Bell-style

If this parameter is set to “Yes”, user will be able to make Bellcore style 3-way

 

 

3-way Conference

 

conference. *23 will be disabled.

 

 

 

 

 

 

 

 

Offhook

 

This parameter allows a user to configure a User ID or extension number to be

 

 

Auto-Dial

 

automatically dialed upon offhook. Please note that only the user part of a SIP

 

 

 

 

address needs to be entered here. The HT-386 will automatically append the

 

 

 

 

“@” and the host portion of the corresponding SIP address.

 

 

 

 

NOTE: Please write down the IP address of the ATA if you use this feature as

 

 

 

 

it will disable the IVR and the only way to access the HT-386 is via web

 

 

 

 

configuration page.

 

 

 

 

 

 

 

 

Proxy-Require

 

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

 

 

 

 

 

 

 

Disable Call Waiting

 

Default is No. User can use * code to use this feature per call basis.

 

 

 

 

 

 

 

 

 

 

 

 

 

NAT Traversal

 

This parameter defines whether the HT-386 NAT traversal mechanism will be

 

 

 

 

activated or not. If activated (by choosing “Yes”) and a STUN server is also

 

 

 

 

specified, then the HT-386 will behave according to the STUN client

 

 

 

 

specification. Under this mode, the embedded STUN client inside the HT-386

 

 

 

 

will attempt to detect if and what type of firewall/NAT it is sitting behind through

 

 

 

 

communication with the specified STUN server. If the detected NAT is a Full

 

 

 

 

Cone, Restricted Cone, or a Port-Restricted Cone, the HT-386 will attempt to

 

 

 

 

use its mapped public IP address and port in all of its SIP and SDP messages.

 

 

 

 

If the NAT Traversal field is set to “Yes” with no specified STUN server, the

 

 

 

 

HT-386 will periodically (every 20 seconds or so) send a blank UDP packet

 

 

 

 

(with no payload data) to the SIP server to keep the “hole” on the NAT open.

 

 

 

 

 

 

 

 

Preferred Vocoder

 

The HT-386 supports 6 different codec types including :

 

 

 

 

 

G.711 A/UlawG.723.1, G.726, G.729A/B, iLBC.

 

 

 

 

 

A user can configure codecs in a preference list that will be included with the

 

 

 

 

same preference order in SDP message.

 

 

 

 

 

 

 

 

Voice Frames per TX

 

This field contains the number of voice frames to be transmitted in a single

 

 

 

 

packet. When setting this value, the user should be aware of the requested

 

 

 

 

packet time (used in SDP message) as a result of configuring this parameter.

 

 

 

 

This parameter is associated with the first codec in the above codec

 

 

 

 

Preference List or the actual used payload type negotiated between the 2

 

 

 

 

conversation parties at run time.

 

 

 

 

 

e.g., if the first codec is configured as G723 and the “Voice Frames per TX” is

 

 

 

 

set to be 2, then the “ptime” value in the SDP message of an INVITE request

 

 

 

 

will be 60ms because each G723 voice frame contains 30ms of audio.

 

 

 

 

Similarly, if this field is set to be 2 and if the first codec chosen is G729 or

 

 

 

 

G711 or G726, then the “ptime” value in the SDP message of an INVITE

 

 

 

 

request will be 20ms.

 

 

 

 

 

If the configured voice frames per TX exceeds the maximum allowed value,

 

 

 

 

the HT-386 will use and save the maximum allowed value for the

 

 

 

 

corresponding first codec choice. The maximum value for PCM is 10(x10ms)

 

 

 

 

frames; for G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames;

 

 

 

 

for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively. Please be

 

 

 

 

careful when massage those parameters.

 

 

 

 

 

 

 

 

G723 Rate:

Encoding rate for G723 codec. By default, 6.3kbps rate is set.

 

 

 

 

 

 

 

iLBC frame size:

 

iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might need

 

 

 

 

to be set.

 

 

 

 

 

 

 

 

iLBC payload type:

 

Payload type for iLBC. Default value is 97. The valid range is between 96 and

 

 

 

 

 

 

 

 

Grandstream Networks, Inc.

HT-386 User Manual

Page 24 of 34

 

 

 

 

Firmware 1.0.3.64

Last Updated: 2/2007

 

Image 24
Contents Grandstream Networks, Inc Table of Contents Table of Figures Welcome Installation Button and two colors led indicatorWAN Product Overview DhcpFeature Keys Audio FeaturesFeatures Lines/SIP AccountsUniversal Switching Power AdaptorDimension WeightBasic Operations Main MenuReset ExamplesDirect IP Calls Attended Transfer Star Code Style 3-way ConferenceBellcore Style 3-way Conference Pstn Pass Through Call Features Button flashes every 2 seconds Red light steadyButton flashes at 1/10 second Button flashes every secondConfiguration Guide Dhcp ModeUser Level Password Web pages allowed MAC Address IP AddressProduct Model Software VersionPassword End UserWeb Port Dhcp hostnamePstn Access Code Admin PasswordNo Key Entry timeout Firmware UpgradeGrandstream Networks, Inc Authenticate ID AuthenticationUnregister On Reboot SIP ServerEnable Call Features Disable Call WaitingSend Flash Event Use Bell-styleFax Mode Lock keypad updateSpecial Feature Volume AmplificationScreenshot of Configuration Update Mode Call Progress TonesGrandstream Device Configuration Software Upgrade IVR MethodTftp Server Downloading Directions Restore Factory Default Setting Glossary of Terms Grandstream Networks, Inc Grandstream Networks, Inc Grandstream Networks, Inc
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HT386 specifications

Grandstream Networks has established itself as a leading provider of communication solutions, specializing in voice-over-IP (VoIP) technology. Among its diverse product line, the HT386, HT496, HT488, HT486, and HT286 analog telephone adapters stand out as exceptional devices tailored for seamless integration into modern telecommunication systems.

The Grandstream HT386 is particularly noted for its robust performance and versatility. It supports up to 4 lines, making it ideal for small to medium-sized businesses that require efficiency and reliability. The unit features advanced security protocols such as SRTP and TLS to protect voice communications, ensuring that data is secure during transmission. Additionally, the HT386 boasts an easy installation process and web-based management, which simplifies configuration and maintenance.

Next in line is the HT496, which caters to users with even more demanding requirements. This adapter accommodates up to 4 FXS ports, allowing the connection of multiple analog devices. Enhanced features like 2 SIP accounts and high-definition voice codecs ensure clear audio quality. The HT496 is designed to offer seamless interoperability with various IP routers and switches, making it a flexible solution for businesses expanding their communication framework.

The HT488, another notable entry, is geared towards those looking for high-performance analog telephony. With support for 2 lines and advanced echo cancellation technologies, it guarantees crystal-clear calls, minimizing disruptions during conversations. Additionally, it provides multiple network connectivity options, including DHCP and static IP, allowing users to choose the best configuration suitable for their network environment.

The HT486 offers similar benefits but is optimized for users who require a compact solution. This model features an elegant design while maintaining support for essential VoIP features. With 2 FXS ports and built-in firewall capabilities, it ensures secure and efficient communication for residential and small business users.

Finally, the HT286 is perfect for those seeking an entry-level adapter without compromising on quality. Supporting a single line with a straightforward setup process, it is ideal for users transitioning from traditional phone systems to VoIP technology. This model is also compatible with various VoIP service providers, ensuring users have flexibility when choosing their phone services.

In summary, Grandstream’s HT series—HT386, HT496, HT488, HT486, and HT286—delivers a comprehensive range of features and technologies suited for different communication needs. Each model combines quality with user-friendly interfaces, ensuring that users can fully leverage the benefits of VoIP, whether for personal or business use.